The problem of reduced speech intelligibility of persons with normal hearing and in particular hearing impaired persons under adverse listening conditions, such as restaurants, traffic and other noisy environments, is well known.
Efforts have been made to improve this situation for users of hearing aids, as a number of techniques based on single- and multi-microphone systems have been applied to suppress unwanted background noise.
Single-microphone systems have utilised directional microphones and/or signal filtering, e.g. spectra-filtering, in order to reduce the background noise in relation to the desired signal, i.e. the speech signal.
Multi-microphone systems using fixed beam-forming have been proposed, where the incoming sound can be sampled spatially, and the direction of arrival can be used for discriminating desired from undesired signals. With these systems it is possible to suppress stationary and non-stationary noise sources independently of their spectra. However, in order to achieve an effective cancelling of the undesired signals, the size of the microphone array will be considerably larger than the average size of commonly used hearing aids, e.g. behind-the-ear (BTE) or in-the-ear (ITE) hearing aids.
Multi-microphone systems using adaptive noise cancelling have also been proposed. In these proposed noise cancelling systems adaptive noise cancellation is used to try to null out the interfering noise source or sources. An example of such a system is disclosed in Journal of the Acoustical Society of America, 103 (6), June 1998, pp. 3621-3626, J. Vanden Berghe and J. Wouters:“An adaptive noise canceller for hearing aids using two nearby microphones”.
In the above article a noise cancelling system using two nearby microphones is described, which system contains two sections, in which the signals are processed. The signals from the microphones, which contain both noise and speech, are led to the first section, which serves to generate a speech reference signal and a noise reference signal. These reference signals are led to the second section, which produces an output signal, in which the noise has been reduced in relation to the speech signal. Each section in this system comprises an adaptive filter. The first section in this piece of related art comprises an adaptive filter, the output of which is intended to converge towards a delayed signal from the primary microphone, while the adaptive filter in the second section of the system models the difference between the noise reference and the delayed speech reference, and subsequently the noise portion in the delayed speech is subtracted.